This thesis details the design of a selection operator used in a Genetic Algorithm. The Genetic Algorithm is used for loudspeaker filter design of three way loudspeakers for which tournament selection was chosen as selection operator. A methodology is proposed and used to tune the parameters of tournament selection, which is based on diversity and fitness of the population. Besides basic tournament selection, two new adaptive selection operators based on tournament selection are proposed to improve its functionality. The first adaptive selection operator uses noise proportional to the fitness variance of the population to improve the efficiency of the genetic algorithm. The second adaptive selection operator uses a convergence stage to speed up the convergence towards the optimal filter. After the presented tuning process in this thesis, the latter adaptive selection operator was found to perform better. The optimal selection operator and parameters found in this thesis will not translate to every application, because they heavily depend on the design and the application of the genetic algorithm. However, the presented comparison of selection operators, the provided performance metrics and design methodology can still be used to guide the choice and the tuning process of a selection operator used in any genetic algorithm.

This paper reports the design of a part of a genetic algorithm, which is made to design analog filters for loudspeakers. The part in this report is the part which deals with the representation of electrical filter circuits, the mutation of these filters, and finding their transfer function. The considered representations are graph coding and a tree data structure. They are compared on intuitiveness, how well mutations can be performed, and the complexity of calculating the transfer function. The tree structure is reasoned to be the most suitable. Described is which mutations can be performed on a filter by the final program, as well as how embryo circuits are made, how the transfer function is calculated, and which hyperparameters were designed and how they were set. Finally, the design is implemented using Python and the operations are tested.

This thesis describes the design and implementation of a controller and GUI for an AI loudspeaker filter design program. This program uses a genetic algorithm to create a combination of analog passive filters, one for each driver in a loudspeaker system. In the controller, two cost functions are designed to evaluate these filter combinations, and when the genetic algorithm has created its final filters, unnecessary components are removed and all component values are optimized. A GUI is created to allow easy user interaction.
For selecting a cost function, not enough data was obtained to make a deliberate decision based on performance. Therefore, this decision was based on theory and subordinate features. Nonetheless, the final program is able to create flat acoustic responses in a margin of 2 dB.

Global Navigation Satellite Systems (GNSSs) have become a critical part of the infrastructure of modern society. Radio interference can introduce position or timing errors in systems that use GNSS or, in a worst-case scenario, block the reception of GNSS signals in full. Part of this critical infrastructure is, among others, power plants, banks, and transport. Interference of GNSS signals could originate from nature, such as solar activity or ionospheric effects. Other interference could originate from unintentional sources (e.g., radio signals from a malfunctioning radio tower) or intentional sources such as a jamming or spoofing device. The latter is what this thesis will focus on. This thesis consists of two parts. The first part is about jamming and elaborates on the impact of seven different forms of jamming on two types of GNSS receivers, a time-worn receiver and a cuttingedge receiver. The cutting-edge receiver has as option to turn on Interference Mitigation (IM). The performance of both receivers is the roughly the same in case the IM is turned off on the cutting-edge receiver. However, when the IM is turned on the cutting-edge receiver clearly is more resillient to the jamming signals. In the second part of the thesis various types of spoofing are discussed. Due to time and hardware restrictions it was not possible to perform synchronous spoofing, which is an advanced form of spoofing. Instead, various concepts are discussed that describe how synchronous spoofing could be achieved.

This thesis covers two topics. The first one is signal design for accurate Time-of-Arrival estimation using a number of frequency separated signals. Rather than use a full UWB band, we will use sparse subband signals spanning the full band to construct a new virtual UWB signal. To evaluate the performance of the constructed signal, Cramer-Rao lower bound and auto-correlation are used. And given a given fixed bandwidth the number of subbands within a 1 GHz UWB channel, and optimal subbands' allocation will be found based on the evaluation results. Our results show that when three 50 MHz subbands are used to construct a virtual 1GHz UWB signal, a lower CRLB and better auto-correlation performance can be reached when subbands are close to the edges of the virtual band. However, the auto-correlation still has multiple peaks, which poses a serious challenge for accurate time estimation.The second topic is to investigate the frequency dependence of the channel impulse response of subbands with different frequency separations. We propose a covariance calculation method to determine the frequency dependence which changes with frequency separation.To validate the method, different artificial UWB channels with distinct paths are given. The results show that covariances between the subband CIRs stay at a high level when measured at the direct path and the majority of interference caused by other paths can be eliminated by a wider bandwidth subband. Given UWB channels measured from 5 to 10 GHz with a link-budget of 120dB, the frequency dependence of the direct path and reflections are determined, different bandwidths and frequency separations are used, and the results show that the channel impulse response of the subbands will become different when measured at different center frequencies, where the difference increases with an increased frequency separation of the subbands. The correlation of the direct path is maintained over larger frequency distances than that of reflected paths.

The displacement of a loudspeaker cone is not linearly proportional to the voltage/current at its input provided. This is due to electrical and mechanical limitations. The distortion is more noticeable at lower frequencies. This report is one of two reports, that will explain how to minimize this distortion. The motion of the loudspeaker cone will be measured and compared with the input voltage. This way the error in the output can be found. Finally, a system will adjust the current through the voice coil to decrease this error. An accelerometer is used as a sensor and the ADAU1777 is used to compensate for the error in output displacement. Finally, a voltage to current converter is used to convert and amplify the output voltage from the ADAU1777 into a current signal to drive the loudspeaker. This report focuses on the voltage to current amplifier. To begin the design, a problem definition and program of requirements will be given. Usually, loudspeakers are voltage driven. This report begins by first explaining why a current-driven loudspeaker delivers less distortion. Then an attempt is made to build a single operational amplifier (opamp) design that meets the requirements for a real load. For the single opamp design, a simple version of the operational transconductance amplifier (OTA) will be compared against a Howland model based on noise performance and circuit analysis, where it will be shown that the simple OTA has a larger signal to noise ratio and does not need to meet any additional criteria to maintain a high output impedance. However, the simple OTA still does not meet the Signal to Noise (SNR) . The simple OTA also did not meet the maximum Total Harmonic Distortion (THD) at 800 Hz. Then, a two opamp design will be constructed using a composite configuration to increase the SNR and decrease the THD. This design does meet the requirements given, at least for a purely resistive load. Then an electrical model of a loudspeaker, which has a complex impedance, will replace the resistive load. Now the composite design for the resistive load needs to be adjusted for the inductive characteristic of a loudspeaker at higher frequencies. The resulting design has a THD of 0.000427\%, a PM of 51$\degree$ and a SNR is 100.92 dB. All results will be given as simulations, because the current pandemic (COVID-19) does not make it possible to physically construct and measure this model. Micro-Cap 12 is used for circuit simulations and Slicap for circuit noise analysis.

This thesis report focuses on possible methods of digital implementation of motional feedback in a bass loudspeaker. In this thesis, the control system will be created for a monopole and a dipole speaker specifically. It does so by sketching the outlines of such a system and examining possible designs for the components required to suppress distortion which is mainly created by the speaker. This thesis shows that the digital implementation indeed allows filtering with very little latency but there are limitations in accuracy due to the conversion from the analogue to the digital domain. It is shown in this thesis that a partly analogue and partly digital system is desired for better error correction. A desired input-output gain of 30=29.54dB is achieved. The controller that is suggested is a controller based on the transfer function of the speaker. By transforming the input signal with the inverse of this open-loop transfer, the closed-loop transfer will be flattened. The instrumentation amplifier used to subtract the feedback signal from the input signal is designed such that the influence of the quantization noise is minimized. Loop-gain is added by the instrumentation amplifier, the controller and voltage-to-current amplifier. By creating loop-gain as much as possible using the digital implementation, noise is suppressed from 30dB up to 90dB within the range of frequencies of interest. The designed components in the motional feedback system allow a loop-gain up to 70dB and an input to output voltage gain of 30dB before instability occurs.

The aim of this research is to create a nonlinear model of a loudspeaker to analyze the open-loop distortion as well as the closed loop performance with linear and nonlinear controllers. A method is proposed for measuring the dominant nonlinear parameters of a loudspeakers. Furthermore, the loudspeaker distortion is both measured experimentally and simulated using the nonlinear model. The method for nonlinear system identification suffers from poor accuracy and takes into account neither the Eddy current losses, frequency dependent compliance and damping nor visco-elastic effects of the loudspeaker surround material. The simulations and measurements of distortion are not in agreement.

This thesis is part of the Bachelor Graduation Project of Electrical Engineering at Delft University of Tech-nology. This contains the complete design process of an analogue implementation of a Motional Feedbackcontroller. Although the complete controller has been designed, no significant test results have yet beenacquired. However, theoretically the linear distortion at20H zis reduced by99.88%by the controller. Alsois predicted that the nonlinear distortion is suppressed, because of the high loop gain. The feedback signalis generated using an accelerometer. The controller mainly consists of a PI-controller and a predistortionfilter by means of a Linkwitz Transform.

This thesis describes the digital implementation of a motional feedback system for a bass loudspeaker. Motional feedback is used to suppress the linear and non-linear distortions produced by the loudspeaker, especially at the low frequencies. An accelerometer is mounted on the cone of the loudspeaker to provide the feedback signal. The controller which consists of a PI controller and an equalizer are implemented on an FPGA. The equalizer, which is the inverse of the linear model of the loudspeaker, is used to compensate for the linear distortion. The PI controller with negative feedback is used to suppress the non-linear distortion. Not all measurement results are available at the moment of submission of this thesis. However, simulations were carried out on the model of the loudspeakers which show that the linear distortion is fully suppressed. The reduction of the non-linear distortion due to the controller can not be seen in the simulations.

A phase-domain analog-to-digital converter (PhADC) is a promising alternative to a pair of amplitude-domain in-phase and quadrature (IQ) ADCs for low power FSK/PSK demodulation, but due to the nonlinear amplitude-to-phase conversion, IQ offsets and gain mismatch can produce nonlinear phase distortions which that can lead to phase errors and an increase in bit-error rate (BER).
An IQ offset and gain mismatch detection technique is developed for PhADCs and verified. The offset and gain mismatch is estimated by detecting the phase vector distribution imbalance among the four phase quadrants after the analog-to-phase conversion. A feedback path has been constructed between the PhADC output and the offset/mismatch compensation interface. The closed loop will help the compensation interface to settle to the proper compensation values of the offset and mismatch. Incorporating this cancellation loop in a receiver system can improve its sensitivity and robustness.
In a conventional IQ ADC receiver we have two quantizations, i.e., one for I and one for Q, and then process the information to extract the phase. By contrast, the PhADC, due to its embedded demodulation attribute, performs only phase quantization. So only one quantization is needed. Because of its compactness, the hardware is simpler and thereby consumes less energy. Moreover the PhADC is immune to magnitude variations, because in an IQ ADC, amplitude quantization noise produces larger phase quantization noise at small vector
magnitudes, while in a PhADC a larger vector has the same phase quantization error as a small vector. Because of these advantages the PhADC is a very good candidate for low power communication. A detection algorithm has been developed that detects the phase vector distribution imbalance between the left and right IQ complex half plane in case of I offset and between the top and bottom of IQ complex half plane in case of Q offset. Using this imbalance we can determine the sign and size of IQ offset. A similar approach has been done for IQ gain mismatch, where the phase vector distribution imbalance is detected between the four phase quadrants in the IQ complex plane. A mixed-signal approach, i.e., detecting the offset and mismatch in the digital domain and compensate in the analog domain.
It is well known that in the IQ ADC receiver the offset and mismatch can be detected and calibrated if necessary before the amplitude is converted into phase in the digital domain, but in the PhADC receiver the offset and mismatch cannot be determined directly due to the absence of amplitude information. Here, we use phase information rather than amplitude information for detection and compensation. For this reason, we have created a direct mismatch and offset detection technique using the output phase signal of the PhADC. The relation between signal-to-noise power ratio (SNR) and its digital counter parts bit-energy-to-noise density (E_b/N_o) and symbol-energy-to-noise density (E_s/N_o) is established and is used to show the effect of IQ offset and gain mismatch on the BER as a function of channel and phase quantization noise from the PhADC.
It is shown that E_b/N_o and or E_s/N_o are not the same as SNR and less understood ratios that are often confused with SNR. The relation between SNR and Eb/No, or for higher dimensions (i.e., multiple bits per symbol), the E_s/N_o depends on the modulation parameters. For π/4 DQPSK modulation, the following relationship holds:E_b/N_o = SNR + 0.97 dB.
An ideal PhADC receiver for π/4 DQPSK demodulation, with a noise-free channel, can tolerate a maximum IQ offset of 28 mV and a gain mismatch of ± 25 dB for a required BER of 10^-5 according to the IEEE802.15.6 standard. But with a channel noise of SNR= 21dB these become 22 mV and ± 6 dB, respectively. A practical PhADC receiver with the same channel noise can tolerate a maximum IQ offset of 14mV and a gain mismatch of ± 3 dB.
An approach is presented to convert the PhADC receiver analog channel select filter to a digital one for discrete modeling purposes in Matlab Simulink. First, the Laplace transfer functions of the filter stages and from those the overall Laplace transfer function of the total filter are derived. A mapping procedure is proposed to convert the continuous time domain Laplace transfer function to a discrete one.

This thesis project researches the effect of the optimisation time interval on the performance of a self-optimised mobile network. The goal of the thesis is to ascertain if there exists an optimal time interval for the self-optimisation of the KPN network, and what that interval is. In order to research this question, the project uses data from the KPN network as input, and sets up a simulation study in MATLAB. Two areas in the Netherlands are considered in this study – Friesland and Purmerend. The self-optimisation of the network is carried out through the modification of three optimisation parameters – antenna tilt, RS power, and Cell Individual Offset. The scope of the study is limited to LTE in the downlink, for the 800 MHz band. The bandwidth used in this study is 10 MHz. The performance of the mobile network has been studied using KPIs such as 10th throughput percentile, coverage failure rate, call drop rate, and load. In the end, the study analyses the results for each area, for the self-optimisation carried out by modifying the three parameters over several different optimisation time intervals, and discusses their impact on the performance of the network. A comparison has also been drawn between the performance of a self-optimised network and an un-optimised network, to highlight the gains achieved with SON. Finally, recommendations are made regarding a suitable time interval, and a relative comparison between suitability of the three optimisation parameters has been drawn.
The study finds that a suitable time interval for optimisation does exist, and is 240 minutes, for both the simulation areas. The study finds RS power to be the most suitable parameter for self-optimisation, in both the areas. However, the research runs into some unexpected results with respect to the optimisations using tilt angle, and has been discussed in detail in the report. Significant gains are observed with SON, as compared to the case of ‘No SON’ or an un-optimised network.